On Thu, Oct 21, 2010 at 02:46:16PM +0530, Jigar Joshi wrote:
Here I am expecting to be configured following scenario:
User calls : it will play a sound will ask for input DTMF, then call will be
given to particular extension for any DTMF entered.
But its not working as expected.
I have
On 21 Oct 2010, at 10:16, Jigar Joshi wrote:
I have attached the dial plan file.
In what format?
S
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On 09/06/2010 10:28 AM, Olivier wrote:
Hi,
With a 1.4.35 or 1.6.1.19, I'm facing this behaviour :
- extension 7002 is a SIP hard phone currently configured to forward
incoming calls to extension 7003, when a call is unanswered within a 10s
time frame
- when extension 7001 is calling
2010/9/7 Kevin P. Fleming kpflem...@digium.com
On 09/06/2010 10:28 AM, Olivier wrote:
Hi,
With a 1.4.35 or 1.6.1.19, I'm facing this behaviour :
- extension 7002 is a SIP hard phone currently configured to forward
incoming calls to extension 7003, when a call is unanswered within a
Hi,
With a 1.4.35 or 1.6.1.19, I'm facing this behaviour :
- extension 7002 is a SIP hard phone currently configured to forward
incoming calls to extension 7003, when a call is unanswered within a 10s
time frame
- when extension 7001 is calling extension 7002 with a Dial(SIP/7002,20)
statement
Hello,
I have used the Dial option 'r' before in older Asterisk versions and it
behaved as expected, that is, Asterisk would always give ringback audio
before the call was answered no matter what.
It has been a while that I have used version 1.4.33.1 without any the
'r' option. Now that I
Hi!
Does anyone know if the behavior of 'r' has changed but was not
documented? If yes, then how does one inject ringback audio before the
call is answered on the called end?
Search this list for progress or progressinband, and look at the voip-
info wiki.
Please note that I don't claim myself a guru, just happened to be working with
Asterisk for some good number of years, so probably know some stuff better than
others.
As for the number of lines, 1800 lines will come down to 1000 lines using AEL
but not the opposite.
When I'll be back home,
Hi,
I can't figure out what syntax to use with the Dial() M parameter
for the AEL parser to interpret properly. Creating an AEL
macro named macro-screen() partly works as a hack, but it must
not turn into a gosub properly, so I get warnings about the return;.
Dial(...,tgM(screen)) with the
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark G. Thomas
Subject: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2
Hi,
I can't figure out what syntax to use with the Dial() M parameter
for the AEL parser to interpret properly
On Tuesday 03 August 2010 13:19:11 Mark G. Thomas wrote:
I can't figure out what syntax to use with the Dial() M parameter
for the AEL parser to interpret properly. Creating an AEL
macro named macro-screen() partly works as a hack, but it must
not turn into a gosub properly, so I get warnings
Hi,
On Tue, Aug 03, 2010 at 01:49:11PM -0500, Tilghman Lesher wrote:
On Tuesday 03 August 2010 13:19:11 Mark G. Thomas wrote:
I can't figure out what syntax to use with the Dial() M parameter
for the AEL parser to interpret properly. Creating an AEL
macro named macro-screen() partly works
AEL is very simple and the instructions on voip-info.org are enough to learn
it. In fact I can't understand how can one write complex dial plans not
using AEL, you simply can't do it using standard format used in
extensions.conf.
As for the tutorials, there is no specific website for them as per
AEL is very simple and the instructions on voip-info.org are enough to learn
it. In fact I can't understand how can one write complex dial plans not using
AEL, you simply can't do it using standard format used in extensions.conf.
As for the tutorials, there is no specific website for them as
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
unsero...@aol.com
Subject: Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2
AEL is very simple and the instructions on voip-info.org are enough to
learn it. In fact I can't
wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *unsero...@aol.com
*Subject:* Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2
AEL is very simple and the instructions on voip-info.org are enough to
learn
On Tue, Aug 3, 2010 at 1:30 PM, Mark G. Thomas m...@misty.com wrote:
I didn't know there was a U option. I don't see any mention of it
on the voip-info.org wiki or other Dial() documentation, but didn't
check for new options in the built in documentation until just now.
I updated the dial
What determines how long SIP channel waits, when you dial a peer with no
registration, before returning ${DIALSTATUS} CONGESTION?
When I dial a peer with no registration, SIP channel currently waits
many seconds before returning ${DIALSTATUS} CONGESTION - how can I
shorten this timeout?
--
When I dial a peer with no registration, SIP channel currently waits
many seconds before returning ${DIALSTATUS} CONGESTION - how can I
shorten this timeout?
Look at qualify=yes for that peer.
Use ChanIsAvail() before you dial.
Use SIPPEER(peername|status) to check registration status.
Use
: Re: [asterisk-users] Dial options not working
Thanks, but I don't have any *dahdi*.conf file here... (I check in
/etc/asterisk)
--
Anahi Ludueña
Hi, do you mean what kind of extension I have? it is SIP, but from it,
everything works well...
In the SIP extension, the DTMF mode is rfc2833.
Thanks,
From: asteriskus...@dovid.net
To: asterisk-users@lists.digium.com
Date: Wed, 30 Jun 2010 13:54:50 +0300
Subject: Re: [asterisk-users] Dial
PM
Subject: Re: [asterisk-users] Dial options not working
Hi, do you mean what kind of extension I have? it is SIP, but from it,
everything works well...
In the SIP extension, the DTMF mode is rfc2833.
Thanks,
From: asteriskus...@dovid.net
To: asterisk-users@lists.digium.com
Date
Hi, yes, I've just tried to use the dtmf mode inband, but it doesn't work with
landline phones or cell phones...
Thanks,
Anahi Ludueña
Date: Wed, 30 Jun 2010 12:56:59 +0100
From: kwat...@geniusgroupltd.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dial options
Hi, I have an extension which has the follow me option activated. The followme
option should go to a IVR if no answer...
The problem that I have is that everything works when I'm calling it from my
extension, but if I use any landline phone or a cell phone, I'm unable to enter
any options.
29, 2010 4:51 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial options not working
Hi, I have an extension which has the follow me option activated. The
followme option should go to a IVR if no answer...
The problem that I have is that everything works when I'm calling
Thanks, but I don't have any *dahdi*.conf file here... (I check in
/etc/asterisk)
Anahi Ludueña
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 29 Jun 2010 16:54:01 -0500
Subject: Re: [asterisk-users] Dial options not working
Check your DTMF
{\rtf1\ansi\ansicpg1252\fromhtml1 \fbidis \deff0{\fonttbl
{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
\uc1\pard\plain\deftab360 \f0\fs24
{\*\htmltag19 html
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
Where did you define EXTERN?
S
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Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, April 28, 2010 12:57:54 AM
Subject: Re: [asterisk-users] Dial plan question.
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip
Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, April 28, 2010 12:57:54 AM
Subject: Re: [asterisk-users] Dial plan question.
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
Where did you define EXTERN
Try: exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20) ?
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New to Asterisk? Join us for a live introductory webinar every Thurs:
Thanks a lot Jim and Ryan.
It worked with changing the order as you suggested.
--
Few more questions on Dial plan:
use case:
when some one in my pbx calls 100.200, I have translations well defined, Media
also (media via asterisk) --Works.
when some one calls bob, or for any names I am adding
On Tue, Apr 27, 2010 at 8:48 PM, Aditya Kumar adityakumar...@yahoo.comwrote:
Hi All,
pl help me with this basic question.
I have a users (soft clients) with usernames having Alphabetics.
I want to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved
Hi All,
pl help me with this basic question.
I have a users (soft clients) with usernames having Alphabetics.
I want to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved .
so How can I make the configuration to work ( with numbers I can get this done
I am not sure what your problem is. You can have a numeric extension dial an
alphabetic sip user.
exten = 123,1,Dial(SIP/somename)
The soft phone registers to your box with whatever username you set up.
If your phone can dial alpha then you can have
exten = alpha,1,Dial(SIP/$(EXTEN})
--
u pl give me complete
numbering plam
From: Jim Dickenson dicken...@cfmc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tue, April 27, 2010 7:09:45 PM
Subject: Re: [asterisk-users] Dial plan question.
I am
, April 27, 2010 10:11:16 PM
Subject: Re: [asterisk-users] Dial plan question.
Thanks a lot jim for the reply.
My issue is :
there is no numbers involved. I have soft clients.
when a user (bob) calls Alex,
he just opens his sip client and types in a...@pbx.com
so how can I write the translations
Hi people, I have an extension which has configured the follow me (it derives
to an IVR).
If in my dialplan I put Dial(extenX) (where extenX is that extension) and if it
is not available, it should execute the IVR, is that right?
Well, I think it should be, but it doesn't...
Here is my CLI:
I want to use Asterisk as a general message delivery system here.
That is, I want to be able to have a (shell, perl, etc.) script on my
Asterisk server dial an extension, wait for it to be answered and then
play a sound file and then hang up, or even wait for a response or
reactions to some IVR.
On Thu, Apr 08, 2010 at 07:00:11AM -0400, Brian J. Murrell wrote:
I want to use Asterisk as a general message delivery system here.
That is, I want to be able to have a (shell, perl, etc.) script on my
Asterisk server dial an extension, wait for it to be answered and then
play a sound file
AGI and AMI is what you need for this.
AMI is for originating the call between extensions
AGI for playing file of your choice.
Both these APIs are well documented
http://www.voip-info.org/wiki/view/Asterisk+AGI
http://www.voip-info.org/wiki/view/Asterisk+manager+API
--
Thanks Regards,
Have a look at the call files examples of voipinfo
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Its not too hard to do what you want
Cheers Duncan
On 8/04/2010, at 11:00 PM, Brian J. Murrell wrote:
I want to use Asterisk as a general message delivery system here.
That is, I
Hello all!
I having some trouble with a OpenVox A1200P card equiped with 5 FXO
and 7 FXS ports, all ok with FXO ports, but the FXS ones are having
some strange problem:
With a telephone connected to any FXS port, when i dial some
extension number on this phone, i receive a busy
- Chandrakant Solanki solanki.chandrak...@gmail.com wrote:
On Fri, Feb 19, 2010 at 12:21 PM, Gopalakrishnaiyer Venugopal-Q16770
venui...@motorola.com wrote:
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to
19 feb 2010 kl. 11.47 skrev --[ UxBoD ]--:
exten == _988XXX.,1,Dial(DAHDI/g1/${EXTEN},20)
UxBoD - you really have to read the security advisory before sending out such
examples on the mailing list. Please go to http://www.asterisk.org now.
Have a nice weekend!
Thanks,
/O
--
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to modify this so that i can dial numbers with more than 10
digits for example like accessing an IVR menu.
Warm Regards
Venugopal G
On Fri, Feb 19, 2010 at 12:21 PM, Gopalakrishnaiyer Venugopal-Q16770
venui...@motorola.com wrote:
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to modify this so that i can dial numbers with more than 10 digits
for
] On Behalf Of
Chandrakant Solanki
Sent: Friday, February 19, 2010 12:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial Plan configuration in asterisk
On Fri, Feb 19, 2010 at 12:21 PM, Gopalakrishnaiyer Venugopal-Q16770
venui...@motorola.com wrote
:
Nice. :-)
Didn't see that, I concede.
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February 06, 2010 12:10 AM
Subject: Re: [asterisk-users] Dial script
On Sat, Feb 06, 2010 at 05:56:43AM -0500, Thomas Perron wrote:
My inquiry is to understand how I could configure a system to do it.
I have since learned that Asterisk has features in the code to do this
(auto dial out, features.conf and .call files.) The 1 example is
a bit extreme but it
Sent: Saturday, February 06, 2010 4:56 AM
Subject: Re: [asterisk-users] Dial script
My inquiry is to understand how I could configure a system to do it.
I have since learned that Asterisk has features in the code to do this
(auto dial out, features.conf and .call files.) The 1 example
On Sat, 6 Feb 2010, Thomas Perron wrote:
My inquiry is to understand how I could configure a system to do it. I
have since learned that Asterisk has features in the code to do this
(auto dial out, features.conf and .call files.) The 1 example is a
bit extreme but it really does not
- From: Thomas Perron thomas.per...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February 06, 2010 4:56 AM
Subject: Re: [asterisk-users] Dial script
My inquiry is to understand how I could configure a system to do
On Sat, 6 Feb 2010, Thomas Perron wrote:
Karl,
[snip]
Your correct, Google has a lot of information.
You may want to go back and learn how to read replies so you can figure
out who said what. Bottom-posting (since it is how most of us humans*
read) will help.
Not once did Karl mention
-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Saturday, February 06, 2010 12:19 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Dial script
On Sat, 6 Feb 2010, Thomas Perron wrote:
Karl,
[snip]
Your correct, Google has a lot of information.
You may want to go back
On Sat, 6 Feb 2010, Michelle Dupuis wrote:
[snip]
Oh wait, the advent of computers has allowed us to conveniently insert
the most recent text at the TOP of a message, to prevent people from
having to reread the same stuff every time.
A: Because we read from top to bottom, left to right.
Here's a .sig from the m...@openbsd list, (which I couldn't resist
top-posting.)
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
A: Top-posting.
Q: What is the most annoying thing in e-mail?
On Sat, Feb 06, 2010 at 10:13:42AM
Thomas,
Yes you can do this. I actually have done this and run it as a
service under the name Meetmecall. I use MSN as the user interface to
record the message, create phone lists of the numbers that has to be
called and to actually schedule and perform the delivery. It is
possible to
Thank you for your interesting comments.
On Sat, Feb 6, 2010 at 4:14 PM, Erik de Wild: Tripple-o
i...@tripple-o.nl wrote:
Thomas,
Yes you can do this. I actually have done this and run it as a
service under the name Meetmecall. I use MSN as the user interface to
record the message, create
On Fri, Feb 5, 2010 at 10:08 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 5 Feb 2010, Thomas Perron wrote:
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses a defined
On Fri, Feb 5, 2010 at 9:54 PM, Thomas Perron thomas.per...@gmail.com wrote:
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses a defined digit, the call gets bridged to me.
There is a
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February 06, 2010 12:10 AM
Subject: Re: [asterisk-users] Dial script
On Fri, 5 Feb 2010, Karl Fife wrote:
Try this:
#rm -rf /
Copycat!
On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd
Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February 06, 2010 12:10 AM
Subject: Re: [asterisk-users] Dial script
On Fri, 5 Feb 2010, Karl Fife wrote:
Try this:
#rm -rf /
Copycat!
On Fri
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses a defined digit, the call gets bridged to me.
--
_
-- Bandwidth and
On Fri, 5 Feb 2010, Thomas Perron wrote:
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses a defined digit, the call gets bridged to me.
Do you mean the dialed numbers are in
Try this:
#rm -rf /
- Original Message -
From: Thomas Perron thomas.per...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, February 05, 2010 8:54 PM
Subject: [asterisk-users] Dial script
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence
Subject: [asterisk-users] Dial script
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses a defined digit, the call gets bridged to me
On Fri, 5 Feb 2010, Karl Fife wrote:
Try this:
#rm -rf /
Copycat!
On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:
Is there any tested script available for this purpose.
On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org=20
Sure. Add this to root's crontab:
, you must be assumed to be a call
spammer, and you are looking for help in the wrong place.
j
- Original Message -
From: Thomas Perron thomas.per...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, February 05, 2010 8:54 PM
Subject: [asterisk-users] Dial script
Does anyone
Fife karlf...@gmail.com wrote:
Try this:
#rm -rf /
- Original Message -
From: Thomas Perron thomas.per...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, February 05, 2010 8:54 PM
Subject: [asterisk-users] Dial script
Does anyone have a Dial script or a hint
- Original Message -
From: Thomas Perron thomas.per...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, February 05, 2010 8:54 PM
Subject: [asterisk-users] Dial script
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls
Nice. :-)
Didn't see that, I concede.
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February 06, 2010 12:10 AM
Subject: Re: [asterisk-users] Dial script
: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is
it possible?
Hi William,
I appreciate your answer, though can you make things more clear for me:
1- i am not using extensions when registering PBX boxes in IAX files.
2- is inbounx context in the call sender PBX (pbx1) and outbound
@lists.digium.com
Subject: [asterisk-users] Dial multiple extensions and know who
picks up
call
Dear,
I'm currently using a Dial command with multiple destinations and
channels
eg: Dial(SIP/100SIP/101)
I simply would like to know, in real time during the call (from dial
plan or AGI), who has
touati
*Sent:* Thursday, January 28, 2010 3:29 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it
possible?
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that
way:
1) use
[default]
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Friday, January 29, 2010 11:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial cellphone from
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a
cellphone)
my questions now is : am i gonna be able to dial from an IPphone
-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Thursday, January 28, 2010 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it
possible?
Hi Guys,
i am using two PBX's i can call from pbx1
exten = s,1,Answer()
exten = s,n,Background(astcc-please-enter-your)
exten = s,n,Background(zip-code)
exten = s,n,WaitExten(5)
exten = s,n,Read(NUMBER,,5)
exten = s,n,SayDigits(${NUMBER})
exten = 22042,n,Dial(SIP/sipvendor/111,120,A(ginger3))
exten =
Am 17.01.2010 18:39, schrieb Thomas Perron:
exten = s,1,Answer()
exten = s,n,Background(astcc-please-enter-your)
exten = s,n,Background(zip-code)
exten = s,n,WaitExten(5)
exten = s,n,Read(NUMBER,,5)
exten = s,n,SayDigits(${NUMBER})
you might want to add a GoTo(${NUMBER},1)
as well as start
veilen danke timm
cheers
tom
On Sun, Jan 17, 2010 at 2:10 PM, Timm Korte
korte-ast-us...@easycrypt.de wrote:
Am 17.01.2010 18:39, schrieb Thomas Perron:
exten = s,1,Answer()
exten = s,n,Background(astcc-please-enter-your)
exten = s,n,Background(zip-code)
exten = s,n,WaitExten(5)
exten =
Did move 0317998975 phone from my home to my office, didnt need any:
nat=yes in sip.conf, everything worked.
I did also add callcounter=yes in sip.conf so I am not sure how it
will work when I move the phone to my home and need nat=yes again.
Will do some tests later tonight when I am at home.
On
Hi!
Trying to figure out what I am doing wrong...
1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256)
1 Cell phone 00733025975 attached through H323.
extensions.conf
exten = 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1)
exten =
Hello,
What i am trying to do is . Dail a number and ask if you wana talk to
XXX press 1 and if you dont wana talk press any other key.
For this purpose i am using this
linkhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
.
*I am using this option :- *
*M(**x**)*: Executes the macro (x)
Any one have success with Dial M option, Can some one provide an example?
On Thu, Dec 3, 2009 at 12:57 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Hello,
What i am trying to do is . Dail a number and ask if you wana talk to
XXX press 1 and if you dont wana talk press any other
Aah the Problem was i am working on 1.4 and in my mind and logic i was
writing code for 1.6.
The example works perfect
On Thu, Dec 3, 2009 at 3:00 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Any one have success with Dial M option, Can some one provide an example?
On Thu, Dec 3,
Do you have the main-menu sound file in the correct format?
Goksie
On 11/20/09, Steve Edwards asterisk@sedwards.com wrote:
On Fri, 20 Nov 2009, aster...@opensourcesolution.in wrote:
the problem is that when call comes it answers but backgroup main menu
dosent play,for test purpose i had
hi,
i have configured a dial plan in which i need that when a call comes
to extention user should hear main-menu .
version of asterisk
is,
Asterisk 1.6.0.17
my dial plan
/extensions.conf
[internal]
exten
= 2000,1,Dial(Sip/2000)
exten = 2000,2,Answer()
exten =
On Fri, 20 Nov 2009, aster...@opensourcesolution.in wrote:
the problem is that when call comes it answers but backgroup main menu
dosent play,for test purpose i had done
The problem is that you do not have (or have not provided) sufficient
information to solve today's problem.
You should
can you please descibe more in details the bahaviour of your application ?
regards
Mickael
2009/10/19 Anahi Ludueña a_ludu...@hotmail.com
Hi People,
I need to dial an external number, when it is answered, I should digit the
extension.
How can I do that in the DialPlan?
Thanks,
*
Thanks,
I need to make a conference between 2 numbers, one of them is external and it
has an extension. So, I need to dial the number and later enter the extension,
how can I do that?
_
I need to make a conference between 2 numbers, one of them is external and
it has an extension. So, I need to dial the number and later enter the
extension, how can I do that?
something like this :
exten = 5145551212,Dial(Zap/g0/5145556000,20,D(7287))
see
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Charbonneau
Sent: Wednesday, 21 October 2009 1:12 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial a external number with extension
I need to make a conference
Hi People,
I need to dial an external number, when it is answered, I should digit the
extension.
How can I do that in the DialPlan?
Thanks,
Anahi Ludueña
_
¿Sabías que ahora puedes
Hi All,
Would just like to know why it takes so long before asterisk process the
call.
From the time i press send on the phone, it takes about 7 seconds
before i see activity on the asterisk console that asterisk is already
processing the call.
But, if i do a sniff, i immediately see the
: Tuesday, October 13, 2009 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dial Delay
Hi All,
Would just like to know why it takes so long before asterisk process the
call.
From the time i press send on the phone, it takes about 7 seconds
before i
, October 13, 2009 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dial Delay
Hi All,
Would just like to know why it takes so long before asterisk process the
call.
From the time i press send on the phone, it takes about 7 seconds
before
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial Delay
thank you sir. but i think tonedur is only set at zapata.
but my calls are basically SIP and IAX. call from sip user to asterisk
is SIP. then asterisk to another asterisk via IAX. then from
On 14/10/09 4:07 AM, Ron wrote:
Hi All,
Would just like to know why it takes so long before asterisk process the
call.
From the time i press send on the phone, it takes about 7 seconds
before i see activity on the asterisk console that asterisk is already
processing the call.
But, if i
Hi,
documentation shows me:
Dial(Tech/User:passw...@host/Extension,Timeout,Optionen)
This is working for IAX2.
If Iam using
DIAL(SIP/u...@secret@sip.domian.tls/123456)
Asterisk shoes no host with name sip.domian.tls/123456
How to put in extension if using the DIAL command with userid and
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